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Aastra MX-ONE (Ericsson MD110) version 3.2 SP1 to Cisco Unified Communications Manager 7.0 using Cisco Unified Border Element (Enterprise Edition) 1.2 for SIP-to-SIP Interworking
April 6th 2009, Rev# 5
Table of ContentsTable of ContentsTable of ContentsTable of Contents
Introduction..............................................................................................................................................................................................................2 Network Topology....................................................................................................................................................................................................3 Capabilities...............................................................................................................................................................................................................4 Limitations................................................................................................................................................................................................................4 System Components.................................................................................................................................................................................................5
Features....................................................................................................................................................................................................................6 Features Supported..............................................................................................................................................................................................6 Features Not Supported (See limitations)............................................................................................................................................................6
Configuration............................................................................................................................................................................................................7 Configuring the Aastra MX-One.........................................................................................................................................................................7 Configuring the Cisco Unified Communications Manager ...............................................................................................................................12 Configuring Cisco UBE....................................................................................................................................................................................42
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IntroduIntroduIntroduIntroductionctionctionction
• This application note describes the necessary steps and configurations for connectivity between an Ericsson (Aastra) MD110 (MX-One) Version 3.2 SP1 and a Cisco Unified Communications Manager (Cisco UCM) 7.0 using Cisco Unified Border Element (Cisco UBE) 1.2 with SIP trunks.
• The network topology diagram (Figures 1) shows the test setup for end-to-end interoperability between Cisco Unified Communications Manager Release 7.0 connected to the Aastra MX-One (Ericsson MD110) PBX via Cisco UBE 1.2 using SIP trunks. A gigabit Ethernet port is used as the SIP trunking interface. Features tested are basic call, 3-way (ad-hoc) conference, call transfer (attended and unattended), call forward (all, busy and no answer), hold/resume, and DTMF internetworking.
• During testing, a Cisco 3825 voice gateway was used to run the Cisco UBE features set, however other Cisco voice gateways can be used and the decision to choose what Cisco gateway model to use is left to the customer. The customer should choose a Cisco IOS gateway model based on the capabilities and the capacity that will be required based on the planned network deployment. Here is a list of Cisco IOS products capable of running Cisco UBE.
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CapabilitiesCapabilitiesCapabilitiesCapabilities
• Voice calls including supplementary services can be successfully established between phones controlled by the Ericsson and IP phones controlled by the Cisco Unified Communications Manager.
LimitationsLimitationsLimitationsLimitations
• Connected name and connected number cannot be presented during a call. As of the tested software the Aastra MX-ONE PBX does not support P-asserted-Id (PAI) nor Remote-party-Id (RPID) SIP headers.
• During external call transfer, name and number updates are not supported. PAI and RPID are not supported on Aastra PBX, as of the tested software.
• The Aastra PBX does not support external call forward unconditional (Follow-me across SIP trunk).
• A call forward busy or no answer, initiated from a user on the Aastra PBX, includes a History-info SIP header “without” a cause code. The omission of the cause code from the History-info header prevents Cisco UBE from converting the History-info header to a Diversion SIP header. Due to this caveat centralized voice mail integration, using Cisco Unity, Unity Connection or Unity Express, is not possible. Cisco’s UM platforms require the SIP Diversion header for correct functionality.
• DTMF interwork limitation: Cisco UBE translates SIP INFO to RFC2833, but cannot translate RFC2833 to SIP INFO. Since the Aastra PBX supports SIP INFO and does not support RFC2833, DTMF interworking will not work when a Cisco UCM IP phone user attempts to send DTMF towards the Aastra PBX.
• When a user on the Cisco UCM side invokes call hold feature Cisco UCM sends a Re-INVITE with Audio-attribute “inactive” to stop RTP media from flowing. When the call resumes, Cisco UCM sends SDP’less Re-INVITE and requires a 200OK response with SDP fresh offer listing capable codecs with attribute equal to sendrecv. Aastra PBX does not support the SDP’less Re-INVITE and returns a 200K message with Audio-attribute remaining in “inactive” state. The workaround for this limitation is to alter the Audio-attribute which is possible with Cisco UBE using SIP profiles. The intial INVITE sent from Cisco UCM, when the IP phone user places the call on hold, is modified by Cisco UBE at the SDP level so that the “attribute” value is changed from “inactive” to “sendrecv, before the INVITE is forwarded to the Aastra PBX. Using this workaround, when call is RESUMED, voice is established both ways. See configuration section for details.
• Applying the SIP profiles recommended above to address the limitation of SDP’less Re-INVITE during mid-call features causes a corner case failure during a three-way conference that is intitiated by a Aastra side phone user. The three-way conference call is successful but is not able to sustain when the conference initiator drops. One of the SIP profiles needed for MoH to work during a hold is to translate the Audio attribute in response 200 OK from “sendonly” to “sendrecv”. This SIP profile cli should be removed if it is required for the remaining parties of a terminated three-way conference to remain connected, but be aware that Cisco UCM MoH will no longer work. (Scenario: Aastra PBX being the conferencing leader with the other two Cisco UCM conferencing members and then conferencing leader dropping out.). With this SIP profile, voice is established only one way. See configuration section for details.
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Calling name/connected name restrictions > KSCAP:DIR=1251&1252; KEY SYSTEM CATEGORY PRINT DIR TRAF SERV CDIV ROC ITYPE TRM ADC LANG BSEC 1251 03151515 0211120700 011151111 720004 20 0 00100013010000 0 0 1252 03151515 0215120700 011151111 720004 20 0 00100013010000 0 0 END To configure Calling/Connected Name and Number Restricted, use the following command: <KSCAC:DIR=1251&&1252,ADC=00010013010000; To configure Calling/Connected Name and Number Allowed, use the following command: <KSCAC:DIR=1251&&1252,ADC=00100013010000; Call diversion on busy/no reply > CDIDP:DIR=1251&1252; To enable/disable diversion on busy/no reply, use the following command: >CDINI:DIR=1251,DIV=64001; //Call diversion individual number initiate >CDINE:DIR=1251; //Call diversion individual number end
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Configuring the Cisco Unified Communications Manager
1. Cisco Unified Communications Manager Version 2. Region configuration 3. Device pool configuration 4. Media Termination Point registration 5. Conference Bridge registration 6. Media Resource Group List configuration 7. Media Resource Group configuration 8. SIP Trunk Configuration 9. Cisco IP Phone 7960 SCCP Configuration 10. Cisco IP Phone 7971 SIP Configuration 11. Enbloc Route Configuration 12. Calling name and restriction configuration 13. Connected name and number restriction configuration
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Configuring Cisco UBE
CUBE#show version Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS-M), Version 12 .4(22)T, RELEASE SOFTWARE (fc1) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008 by Cisco Systems, Inc. Compiled Fri 10-Oct-08 06:43 by prod_rel_team ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1) CUBE uptime is 5 hours, 27 minutes System returned to ROM by reload at 21:18:11 UTC Mon Apr 6 2009 System image file is "flash:c3825-adventerprisek9_ivs-mz.124-22.T.bin" This product contains cryptographic features and is subject to United States and local country laws governing import, export, transfer and use. Delivery of Cisco cryptographic products does not imply third-party authority to import, export, distribute or use encryption. Importers, exporters, distributors and users are responsible for compliance with U.S. and local country laws. By using this product you agree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws, return this product immediately. A summary of U.S. laws governing Cisco cryptographic products may be found at:http://www.cisco.com/wwl/export/crypto/tool/stqrg.html If you require further assistance please contact us by sending email to [email protected]. Cisco 3825 (revision 1.2) with 225280K/36864K bytes of memory. Processor board ID FTX1227A3CR 2 Gigabit Ethernet interfaces 4 Channelized (E1 or T1)/PRI ports 1 Virtual Private Network (VPN) Module DRAM configuration is 64 bits wide with parity enabled. 479K bytes of NVRAM. 62592K bytes of ATA System CompactFlash (Read/Write) Configuration register is 0x2102 CUBE#show run Building configuration... Current configuration : 3182 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname CUBE ! boot-start-marker boot-end-marker !
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! ! logging message-counter syslog logging buffered 10000000 no logging console enable password cisco ! no Aa new-model ! dot11 syslog ip source-route ip cef ! no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! voice-card 0 dspfarm dsp services dspfarm ! voice service voip address-hiding allow-connections sip to sip redirect ip2ip h323 sip history-info midcall-signaling passthru g729 annexb-all ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! voice class sip-profiles 1 request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"2 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"3 response 200 sdp-header Audio-Attribute modify "sendonly" "sendrecv"3 ! voice translation-rule 5 rule 5 /^5+/ // ! voice translation-rule 6 rule 6 /^6+/ // ! voice translation-profile out-to-CallManager translate called 6 ! voice translation-profile out-to-Ericsson translate called 5 !
2 This SIP profile CLI is required for Call transfer and hold/resume to function properly 3 This command is needed for MoH to work. But enabling this command results in one-way conference when the conferencing phone drops out during the Network/External conference and the phone at the Ericsson end is conferencing.
AOC Advice-of-charge. Information element is sent with the connection setup information for incoming Euro-ISDN connections. The AOC IE is used for call charge calculation.
CUCM Cisco Unified Communications Manager
CCBS Call Completion to Busy Subscriber
CCNR Call Completion on No Reply
CFB Call Forwarding on Busy
CFNR Call Forwarding No Reply
CFU Call Forwarding Unconditional
CLIP Calling Line (Number) Identification Presentation
CLIR Calling Line (Number) Identification Restriction
CNIP Calling Name Identification Presentation
CNIR Calling Name Identification Restriction
COLP Connected Line (Number) Identification Presentation
COLR Connected Line (Number) Identification Restriction
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